audio_soc.c

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/*-
 * Copyright (c) 2019 Oleksandr Tymoshenko <gonzo@FreeBSD.org>
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 * 1. Redistributions of source code must retain the above copyright
 *    notice, this list of conditions and the following disclaimer.
 * 2. Redistributions in binary form must reproduce the above copyright
 *    notice, this list of conditions and the following disclaimer in the
 *    documentation and/or other materials provided with the distribution.
 *
 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR
 * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
 * OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED.
 * IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT,
 * INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT
 * NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
 * DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
 * THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
 * (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF
 * THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 *
 */
 
#include <sys/cdefs.h>
__FBSDID("$FreeBSD$");
 
#include "opt_platform.h"
 
#include <sys/param.h>
#include <sys/systm.h>
#include <sys/bus.h>
#include <sys/clock.h>
#include <sys/kernel.h>
#include <sys/lock.h>
#include <sys/module.h>
#include <sys/endian.h>
 
#include <dev/ofw/ofw_bus.h>
#include <dev/ofw/ofw_bus_subr.h>
 
#include <dev/sound/fdt/audio_dai.h>
#include <dev/sound/pcm/sound.h>
#include "audio_dai_if.h"
 
#define AUDIO_BUFFER_SIZE       48000 * 4
 
struct audio_soc_aux_node {
        SLIST_ENTRY(audio_soc_aux_node) link;
        device_t                        dev;
};
 
struct audio_soc_channel {
        struct audio_soc_softc  *sc;    /* parent device's softc */
        struct pcm_channel      *pcm;   /* PCM channel */
        struct snd_dbuf         *buf;   /* PCM buffer */
        int                     dir;    /* direction */
};
 
struct audio_soc_softc {
        /*
         * pcm_register assumes that sc is snddev_info,
         * so this has to be first structure member for "compatibility"
         */
        struct snddev_info      info;
        device_t                dev;
        char                    *name;
        struct intr_config_hook init_hook;
        device_t                cpu_dev;
        device_t                codec_dev;
        SLIST_HEAD(, audio_soc_aux_node)        aux_devs;
        unsigned int            mclk_fs;
        struct audio_soc_channel        play_channel;
        struct audio_soc_channel        rec_channel;
        /*
         * The format is from the CPU node, for CODEC node clock roles
         * need to be reversed.
         */
        uint32_t                format;
        uint32_t                link_mclk_fs;
};
 
static struct ofw_compat_data compat_data[] = {
        {"simple-audio-card",   1},
        {NULL,                  0},
};
 
static struct {
        const char *name;
        unsigned int fmt;
} ausoc_dai_formats[] = {
        { "i2s",        AUDIO_DAI_FORMAT_I2S },
        { "right_j",    AUDIO_DAI_FORMAT_RJ },
        { "left_j",     AUDIO_DAI_FORMAT_LJ },
        { "dsp_a",      AUDIO_DAI_FORMAT_DSPA },
        { "dsp_b",      AUDIO_DAI_FORMAT_DSPB },
        { "ac97",       AUDIO_DAI_FORMAT_AC97 },
        { "pdm",        AUDIO_DAI_FORMAT_PDM },
};
 
static int      audio_soc_probe(device_t dev);
static int      audio_soc_attach(device_t dev);
static int      audio_soc_detach(device_t dev);
 
/*
 * Invert master/slave roles for CODEC side of the node
 */
static uint32_t
audio_soc_reverse_clocks(uint32_t format)
{
        int fmt, pol, clk;
 
        fmt = AUDIO_DAI_FORMAT_FORMAT(format);
        pol = AUDIO_DAI_FORMAT_POLARITY(format);
        clk = AUDIO_DAI_FORMAT_CLOCK(format);
 
        switch (clk) {
        case AUDIO_DAI_CLOCK_CBM_CFM:
                clk = AUDIO_DAI_CLOCK_CBS_CFS;
                break;
        case AUDIO_DAI_CLOCK_CBS_CFM:
                clk = AUDIO_DAI_CLOCK_CBM_CFS;
                break;
        case AUDIO_DAI_CLOCK_CBM_CFS:
                clk = AUDIO_DAI_CLOCK_CBS_CFM;
                break;
        case AUDIO_DAI_CLOCK_CBS_CFS:
                clk = AUDIO_DAI_CLOCK_CBM_CFM;
                break;
        }
 
        return AUDIO_DAI_FORMAT(fmt, pol, clk);
}
 
static uint32_t
audio_soc_chan_setblocksize(kobj_t obj, void *data, uint32_t blocksz)
{
 
        return (blocksz);
}
 
static int
audio_soc_chan_setformat(kobj_t obj, void *data, uint32_t format)
{
 
        struct audio_soc_softc *sc;
        struct audio_soc_channel *ausoc_chan;
 
        ausoc_chan = data;
        sc = ausoc_chan->sc;
 
        return AUDIO_DAI_SET_CHANFORMAT(sc->cpu_dev, format);
}
 
static uint32_t
audio_soc_chan_setspeed(kobj_t obj, void *data, uint32_t speed)
{
 
        struct audio_soc_softc *sc;
        struct audio_soc_channel *ausoc_chan;
        uint32_t rate;
        struct audio_soc_aux_node *aux_node;
 
        ausoc_chan = data;
        sc = ausoc_chan->sc;
 
        if (sc->link_mclk_fs) {
                rate = speed * sc->link_mclk_fs;
                if (AUDIO_DAI_SET_SYSCLK(sc->cpu_dev, rate, AUDIO_DAI_CLOCK_IN))
                        device_printf(sc->dev, "failed to set sysclk for CPU node\n");
 
                if (AUDIO_DAI_SET_SYSCLK(sc->codec_dev, rate, AUDIO_DAI_CLOCK_OUT))
                        device_printf(sc->dev, "failed to set sysclk for codec node\n");
 
                SLIST_FOREACH(aux_node, &sc->aux_devs, link) {
                        if (AUDIO_DAI_SET_SYSCLK(aux_node->dev, rate, AUDIO_DAI_CLOCK_OUT))
                                device_printf(sc->dev, "failed to set sysclk for aux node\n");
                }
        }
 
        /*
         * Let CPU node determine speed
         */
        speed = AUDIO_DAI_SET_CHANSPEED(sc->cpu_dev, speed);
        AUDIO_DAI_SET_CHANSPEED(sc->codec_dev, speed);
        SLIST_FOREACH(aux_node, &sc->aux_devs, link) {
                AUDIO_DAI_SET_CHANSPEED(aux_node->dev, speed);
        }
 
        return (speed);
}
 
static uint32_t
audio_soc_chan_getptr(kobj_t obj, void *data)
{
        struct audio_soc_softc *sc;
        struct audio_soc_channel *ausoc_chan;
 
        ausoc_chan = data;
        sc = ausoc_chan->sc;
 
        return AUDIO_DAI_GET_PTR(sc->cpu_dev, ausoc_chan->dir);
}
 
static void *
audio_soc_chan_init(kobj_t obj, void *devinfo, struct snd_dbuf *b,
        struct pcm_channel *c, int dir)
{
        struct audio_soc_channel *ausoc_chan;
        void *buffer;
 
        ausoc_chan = devinfo;
        buffer = malloc(AUDIO_BUFFER_SIZE, M_DEVBUF, M_WAITOK | M_ZERO);
 
        if (sndbuf_setup(b, buffer, AUDIO_BUFFER_SIZE) != 0) {
                free(buffer, M_DEVBUF);
                return NULL;
        }
 
        ausoc_chan->dir = dir;
        ausoc_chan->buf = b;
        ausoc_chan->pcm = c;
 
        return (devinfo);
}
 
static int
audio_soc_chan_trigger(kobj_t obj, void *data, int go)
{
        struct audio_soc_softc *sc;
        struct audio_soc_channel *ausoc_chan;
        struct audio_soc_aux_node *aux_node;
 
        ausoc_chan = (struct audio_soc_channel *)data;
        sc = ausoc_chan->sc;
        AUDIO_DAI_TRIGGER(sc->codec_dev, go, ausoc_chan->dir);
        SLIST_FOREACH(aux_node, &sc->aux_devs, link) {
                AUDIO_DAI_TRIGGER(aux_node->dev, go, ausoc_chan->dir);
        }
 
        return AUDIO_DAI_TRIGGER(sc->cpu_dev, go, ausoc_chan->dir);
}
 
static int
audio_soc_chan_free(kobj_t obj, void *data)
{
 
        struct audio_soc_channel *ausoc_chan;
        void *buffer;
 
        ausoc_chan = (struct audio_soc_channel *)data;
 
        buffer = sndbuf_getbuf(ausoc_chan->buf);
        if (buffer)
                free(buffer, M_DEVBUF);
 
        return (0);
}
 
static struct pcmchan_caps *
audio_soc_chan_getcaps(kobj_t obj, void *data)
{
        struct audio_soc_softc *sc;
        struct audio_soc_channel *ausoc_chan;
 
        ausoc_chan = data;
        sc = ausoc_chan->sc;
 
        return AUDIO_DAI_GET_CAPS(sc->cpu_dev);
}
 
static kobj_method_t audio_soc_chan_methods[] = {
        KOBJMETHOD(channel_init,        audio_soc_chan_init),
        KOBJMETHOD(channel_free,        audio_soc_chan_free),
        KOBJMETHOD(channel_setformat,   audio_soc_chan_setformat),
        KOBJMETHOD(channel_setspeed,    audio_soc_chan_setspeed),
        KOBJMETHOD(channel_setblocksize,audio_soc_chan_setblocksize),
        KOBJMETHOD(channel_trigger,     audio_soc_chan_trigger),
        KOBJMETHOD(channel_getptr,      audio_soc_chan_getptr),
        KOBJMETHOD(channel_getcaps,     audio_soc_chan_getcaps),
        KOBJMETHOD_END
};
CHANNEL_DECLARE(audio_soc_chan);
 
static void
audio_soc_intr(void *arg)
{
        struct audio_soc_softc *sc;
        int channel_intr_required;
 
        sc = (struct audio_soc_softc *)arg;
        channel_intr_required = AUDIO_DAI_INTR(sc->cpu_dev, sc->play_channel.buf, sc->rec_channel.buf);
        if (channel_intr_required & AUDIO_DAI_PLAY_INTR)
                chn_intr(sc->play_channel.pcm);
        if (channel_intr_required & AUDIO_DAI_REC_INTR)
                chn_intr(sc->rec_channel.pcm);
}
 
static int
audio_soc_probe(device_t dev)
{
 
        if (!ofw_bus_status_okay(dev))
                return (ENXIO);
 
        if (ofw_bus_search_compatible(dev, compat_data)->ocd_data != 0) {
                device_set_desc(dev, "simple-audio-card");
                return (BUS_PROBE_DEFAULT);
        }
 
        return (ENXIO);
}
 
static void
audio_soc_init(void *arg)
{
        struct audio_soc_softc *sc;
        phandle_t node, child;
        device_t daidev, auxdev;
        uint32_t xref;
        uint32_t *aux_devs;
        int ncells, i;
        struct audio_soc_aux_node *aux_node;
 
        sc = (struct audio_soc_softc *)arg;
        config_intrhook_disestablish(&sc->init_hook);
 
        node = ofw_bus_get_node(sc->dev);
        /* TODO: handle multi-link nodes */
        child = ofw_bus_find_child(node, "simple-audio-card,cpu");
        if (child == 0) {
                device_printf(sc->dev, "cpu node is missing\n");
                return;
        }
        if ((OF_getencprop(child, "sound-dai", &xref, sizeof(xref))) <= 0) {
                device_printf(sc->dev, "missing sound-dai property in cpu node\n");
                return;
        }
        daidev = OF_device_from_xref(xref);
        if (daidev == NULL) {
                device_printf(sc->dev, "no driver attached to cpu node\n");
                return;
        }
        sc->cpu_dev = daidev;
 
        child = ofw_bus_find_child(node, "simple-audio-card,codec");
        if (child == 0) {
                device_printf(sc->dev, "codec node is missing\n");
                return;
        }
        if ((OF_getencprop(child, "sound-dai", &xref, sizeof(xref))) <= 0) {
                device_printf(sc->dev, "missing sound-dai property in codec node\n");
                return;
        }
        daidev = OF_device_from_xref(xref);
        if (daidev == NULL) {
                device_printf(sc->dev, "no driver attached to codec node\n");
                return;
        }
        sc->codec_dev = daidev;
 
        /* Add AUX devices */
        aux_devs = NULL;
        ncells = OF_getencprop_alloc_multi(node, "simple-audio-card,aux-devs", sizeof(*aux_devs),
            (void **)&aux_devs);
 
        for (i = 0; i < ncells; i++) {
                auxdev = OF_device_from_xref(aux_devs[i]);
                if (auxdev == NULL)
                        device_printf(sc->dev, "warning: no driver attached to aux node\n");
                aux_node = (struct audio_soc_aux_node *)malloc(sizeof(*aux_node), M_DEVBUF, M_NOWAIT);
                if (aux_node == NULL) {
                        device_printf(sc->dev, "failed to allocate aux node struct\n");
                        return;
                }
                aux_node->dev = auxdev;
                SLIST_INSERT_HEAD(&sc->aux_devs, aux_node, link);
        }
 
        if (aux_devs)
                OF_prop_free(aux_devs);
 
        if (AUDIO_DAI_INIT(sc->cpu_dev, sc->format)) {
                device_printf(sc->dev, "failed to initialize cpu node\n");
                return;
        }
 
        /* Reverse clock roles for CODEC */
        if (AUDIO_DAI_INIT(sc->codec_dev, audio_soc_reverse_clocks(sc->format))) {
                device_printf(sc->dev, "failed to initialize codec node\n");
                return;
        }
 
        SLIST_FOREACH(aux_node, &sc->aux_devs, link) {
                if (AUDIO_DAI_INIT(aux_node->dev, audio_soc_reverse_clocks(sc->format))) {
                        device_printf(sc->dev, "failed to initialize aux node\n");
                        return;
                }
        }
 
        if (pcm_register(sc->dev, sc, 1, 1)) {
                device_printf(sc->dev, "failed to register PCM\n");
                return;
        }
 
        sc->play_channel.sc = sc;
        sc->rec_channel.sc = sc;
 
        pcm_addchan(sc->dev, PCMDIR_PLAY, &audio_soc_chan_class, &sc->play_channel);
        pcm_addchan(sc->dev, PCMDIR_REC, &audio_soc_chan_class, &sc->rec_channel);
 
        pcm_setstatus(sc->dev, "at EXPERIMENT");
 
        AUDIO_DAI_SETUP_INTR(sc->cpu_dev, audio_soc_intr, sc);
        AUDIO_DAI_SETUP_MIXER(sc->codec_dev, sc->dev);
        SLIST_FOREACH(aux_node, &sc->aux_devs, link) {
                AUDIO_DAI_SETUP_MIXER(aux_node->dev, sc->dev);
        }
}
 
static int
audio_soc_attach(device_t dev)
{
        struct audio_soc_softc *sc;
        char *name;
        phandle_t node, cpu_child;
        uint32_t xref;
        int i, ret;
        char tmp[32];
        unsigned int fmt, pol, clk;
        bool frame_master, bitclock_master;
 
        sc = device_get_softc(dev);
        sc->dev = dev;
        node = ofw_bus_get_node(dev);
 
        ret = OF_getprop_alloc(node, "name", (void **)&name);
        if (ret == -1)
                name = "SoC audio";
 
        sc->name = strdup(name, M_DEVBUF);
        device_set_desc(dev, sc->name);
 
        if (ret != -1)
                OF_prop_free(name);
 
        SLIST_INIT(&sc->aux_devs);
 
        ret = OF_getprop(node, "simple-audio-card,format", tmp, sizeof(tmp));
        if (ret == 0) {
                for (i = 0; i < nitems(ausoc_dai_formats); i++) {
                        if (strcmp(tmp, ausoc_dai_formats[i].name) == 0) {
                                fmt = ausoc_dai_formats[i].fmt;
                                break;
                        }
                }
                if (i == nitems(ausoc_dai_formats))
                        return (EINVAL);
        } else
                fmt = AUDIO_DAI_FORMAT_I2S;
 
        if (OF_getencprop(node, "simple-audio-card,mclk-fs",
            &sc->link_mclk_fs, sizeof(sc->link_mclk_fs)) <= 0)
                sc->link_mclk_fs = 0;
 
        /* Unless specified otherwise, CPU node is the master */
        frame_master = bitclock_master = true;
 
        cpu_child = ofw_bus_find_child(node, "simple-audio-card,cpu");
 
        if ((OF_getencprop(node, "simple-audio-card,frame-master", &xref, sizeof(xref))) > 0)
                frame_master = cpu_child == OF_node_from_xref(xref);
 
        if ((OF_getencprop(node, "simple-audio-card,bitclock-master", &xref, sizeof(xref))) > 0)
                bitclock_master = cpu_child == OF_node_from_xref(xref);
 
        if (frame_master) {
                clk = bitclock_master ?
                    AUDIO_DAI_CLOCK_CBM_CFM : AUDIO_DAI_CLOCK_CBS_CFM;
        } else {
                clk = bitclock_master ?
                    AUDIO_DAI_CLOCK_CBM_CFS : AUDIO_DAI_CLOCK_CBS_CFS;
        }
 
        bool bitclock_inversion = OF_hasprop(node, "simple-audio-card,bitclock-inversion");
        bool frame_inversion = OF_hasprop(node, "simple-audio-card,frame-inversion");
        if (bitclock_inversion) {
                pol = frame_inversion ?
                    AUDIO_DAI_POLARITY_IB_IF : AUDIO_DAI_POLARITY_IB_NF;
        } else {
                pol = frame_inversion ?
                    AUDIO_DAI_POLARITY_NB_IF : AUDIO_DAI_POLARITY_NB_NF;
        }
 
        sc->format = AUDIO_DAI_FORMAT(fmt, pol, clk);
 
        sc->init_hook.ich_func = audio_soc_init;
        sc->init_hook.ich_arg = sc;
        if (config_intrhook_establish(&sc->init_hook) != 0)
                return (ENOMEM);
 
        return (0);
}
 
static int
audio_soc_detach(device_t dev)
{
        struct audio_soc_softc *sc;
        struct audio_soc_aux_node *aux;
 
        sc = device_get_softc(dev);
        if (sc->name)
                free(sc->name, M_DEVBUF);
 
        while ((aux = SLIST_FIRST(&sc->aux_devs)) != NULL) {
                SLIST_REMOVE_HEAD(&sc->aux_devs, link);
                free(aux, M_DEVBUF);
        }
 
        return (0);
}
 
static device_method_t audio_soc_methods[] = {
        /* device_if methods */
        DEVMETHOD(device_probe,         audio_soc_probe),
        DEVMETHOD(device_attach,        audio_soc_attach),
        DEVMETHOD(device_detach,        audio_soc_detach),
 
        DEVMETHOD_END,
};
 
static driver_t audio_soc_driver = {
        "pcm",
        audio_soc_methods,
        sizeof(struct audio_soc_softc),
};
 
DRIVER_MODULE(audio_soc, simplebus, audio_soc_driver, pcm_devclass, NULL, NULL);
MODULE_VERSION(audio_soc, 1);